Asterisk Forums

Board started:Mon Apr 04, 2005 11:49 amDays since started:2502Board version:phpbb 3.0.5All times are :UTC - 6 hoursNumber of posts:124326Posts per day:49.68Number of topics:38514Topics per day:15.39Number of users:71652Users per day:28.63Our newest member :neeleshMost users ever online was 328 on Wed Jun 27, 2007 8:01 am

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The old v4 Notifier worked without Outlook, and we used it here mainly to provide easy access to the Switchboard from the tray. We don't use Outlook here, and the v5 Notifier won't install without it. Is there any way to get the basic functionality without Outlook, like we had with v4?

Statistics : Posted by ricknroll • on Thu Feb 09, 2012 6:37 pm • Replies 0 • Views 4

Taking the above dial-plan, would there be a easier way to do what I'm trying to accomplish that wont have the issue with the autoservice not knowing about the outgoing call?

And thanks for the response as well. Gets me looking into different areas.

Statistics : Posted by valentine77 • on Thu Feb 09, 2012 5:24 pm • Replies 2 • Views 25

You really need to attack this at the gateway side.

Obviously you could fix it by configuring Asterisk for dynamic hosts and adding a section to match the international phone number (I'm not sure if it will be happy with the "+"), but if you have a fixed IP address, it is probably more secure to use that than enable registration.

This should have been on Asterisk Support.

Statistics : Posted by david55 • on Thu Feb 09, 2012 4:50 pm • Replies 1 • Views 11

Please contact Digium's technical support department directly by opening a case:

http://www.digium.com/en/users/support_ ... hp?tab=log

or by telephoning us:

+1 256 428 6000

Cheers

Statistics : Posted by malcolmd • on Thu Feb 09, 2012 4:47 pm • Replies 2 • Views 24

Whilst the macro is running, the caller will be connected to the autoservice process, which eats voice frames and lets control frames, like hangup, queue up. Only when control returns to Dial, will the AST_CONTROL_HANGUP frame be actioned. The bridge code also handles them, for established calls. Autoservice doesn't know about the outgoing side of the call, and the macro is running on behalf of the called party.

Going 100% CPU generally means you have a deadlock situation, but one in which Asterisk is polling to gain the lock. See https://wiki.asterisk.org/wiki/display/ ... rADeadlock for what to do with deadlocks.

Statistics : Posted by david55 • on Thu Feb 09, 2012 4:45 pm • Replies 1 • Views 17

The only user error that normally leads to such crashes is leaving obsolete modules in the modules directory, but the crash is normally during the initial module load. You should therefore look at: https://wiki.asterisk.org/wiki/display/ ... +Backtrace

Statistics : Posted by david55 • on Thu Feb 09, 2012 4:36 pm • Replies 1 • Views 20

Save the values to the AstDB with something like
Set(DB(queuename/valueName)=${theValue}))

You can read them with
${DB(queuename/valueName)}

regards,
Tom

Statistics : Posted by tomdemoor • on Thu Feb 09, 2012 2:56 pm • Replies 1 • Views 18

Hello,

I am very new here.. I tried to solve my idea/problem on 2 forums but nobody could help me, that is why I am asking here.
I am trying to get some info for several days (weeks?) if my idea should work..

I am planing to but WNR3500L v2 router with 480mHz proc and 128mb ram with an usb port.
I know that I can install asterisk on it (the router fully supports tomato firmware)

My 1st question is - will I be able to plug a softphone usb into usb port on my router and will it work? I would like to have a softphone working 24/7 working and a router would be a great think to connect it as it is plugged to the internet 24/7 (a laptop or other computer works only a couple of hours a day - I know that it would be easier to work but I would like to try to plug to router - if possible of course).

I know about IP phones which you can plut to rj45 port in router but this is not what I am looking for.. I just wonder if the router could be a 'soft IP gate' for a usb phone

My 2nd question: Is there a posibility to connect a standard analogue telephone with rj11 socked to a rj45 port in the router and to 'tell' installed asterisk on this router that eg LAN port 3 (rs45) where I plugged analogue telephone with a rj11 to rj45 converter that it should communicate to each other? I know that rj11 have to wires and rj45 has got min. 4 wires.. what about the power? rj11 gives power to telephone as well as far as I know?

ps
I found sth like that but I do not really know if this is what I am looking for:
http://www.voip-info.org/storage/users/ ... G_4967.jpg
and
http://www.voip-info.org/wiki/view/Aste ... sys+WRT54G

Regards,
boba

many thanks for even smalles reply,

Statistics : Posted by boba • on Thu Feb 09, 2012 2:53 pm • Replies 0 • Views 11

Hello,

Not necessarily looking for the solution here, but hoping that people can point me towards possible causes to investigate.

Background: new to Asterisk, intermediate with Linux. Running Asterisk 1.8.8.1 on an unslung NSLU2. Using Ekiga softphone on Ubuntu PC with CallCentric as in/out provider. Can make out-going and in-coming calls with no problem. Only two other things I really want to do with this implementation: voicemail and google voice. Right now, stuck on voicemail

Problem: VoiceMailMain causes immediate segmentation fault. No intro beep or greeting; no nothing. All asterisk processes are killed.

Code: -- Executing [1000@my-sip:3] VoiceMailMain("SIP/111-00000002", "9999@default") in new stack
Segmentation fault

VoiceMail (not VoiceMailMain) is working almost correctly, but not quite. It will answer the phone, allow you to record the message, save the message correctly in the right directory, and give "thank you" message. No error messages are given, but asterisk then drops from CLI, (you get kicked back to linux command line) and asterisk is left in an unstable state. "ps -el" shows that some asterisk processes are still running, but it doesn't work properly anymore (i.e. can't make normal SIP calls).

What I've checked so far: I googled some posts which said that the NSLU2 could have problems (crash) with certain sound formats, so I tried "Record" and "Playback" to see if that caused a problem. Turns out that I could record but not playback wav files. Playback of wav files generated some error messages but didn't cause a crash. ulaw was fine both ways. All the announcements seem to work fine in VoiceMail. I even removed all sound files from asterisk to see what would happen, and all the VoiceMail prompts (as expected) disappeared, but otherwise the behaviour was the same. I don't think that this is my issue.

As you probably know, the NSLU2 has very little memory, so some people said that that could cause it to crash, so I turned autoload to off and only loaded a bare minimum of modules (followed a slimming guide on voip-info), but that didn't seem to do anything. Also removed most other processes from NSLU2 so that Asterisk was alone, but VoiceMailMain still crashes immediately on first call. I know that the NSLU2 is short on memory, but I've run apache/mysql on here without much problem, so maybe it's a lack of memory, but I'm not sure.'

This was my best idea. Seems possible/likely that there's a connection between the problem at the end of VoiceMail and the beginning of VoiceMailMain, so I thought maybe there's a function that records a left voice mail event at the end of VoiceMail in a DB or somewhere that was crashing, and then the VoiceMailMain checks the same DB at the beginning of its work and crashes too. I thought CDR seemed like a likely culprit, so I disabled that (I'm 90% sure I did, but I'm still new at asterisk), but same problem.

I'm looking for new things to investigate. Another idea similar to the one above was sendvoicemail. I don't have an email system on the Slug, so I disabled this, but maybe Asterisk/VoiceMail still does some check, even if the sendvoicemail = no, and it's crashing there? Don't know why that would affect VoiceMailMain anyway.
That's it. Like I said, I doubt anyone has the silver bullet, but I'd appreciate any ideas. I really want to get this working and I think it should be doable. Thanks.

Statistics : Posted by allerretour38 • on Thu Feb 09, 2012 2:07 pm • Replies 0 • Views 8

My thoughts, take it or leave it:

Switchvox is a closed appliance. You will not be able to install anything on it, so you will need a separate server if you want to run Hylafax. Concurrent fax licenses on Switchvox will cost ~$40 each. (see: http://store.digium.com/products.php?category_id=165 )

As for fax functionality in Switchvox, read more here: http://www1.digium.com/en/products/swit ... ing-faxing
You may notice in the top paragraph on that page that Switchvox only will support fax over analog or PRI channels. You can try T.38 if you want to, but you will not get any support on doing so.

That said, I think you should stick with Hylafax as your fax solution. You can connect Hylafax to Switchvox over IAX using IAXTel to utilize the same Analog / PRI / SIP trunks and not incur any Switchvox licensing costs to do so (we have done this in the lab, and are planning to roll it out to a couple of our customers soon), and you will have many Desktop faxing options with Hylafax (the Switchvox desktop option is broken in Windows 7, not Digium's fault...) You will be limited by the Switchvox's maximum concurrent call limit - on an AA305 that's 45 calls... if you plan to scale as large as you're thinking, I'd jump up to an AA355 so you can handle 75 concurrent calls / faxes.

Statistics : Posted by ricmarques • on Thu Feb 09, 2012 1:42 pm • Replies 1 • Views 36

Very new to this, but I am trying to do a simple inbound to outbound call

The issue I'm having is this,.. The caller comes into Asterisk, I ask them to record their name and enter the number they wish to call. I then Dial the number they entered with the M option and send the called person to that macro if they answer. The called person is asked a few questions for billing, and if they pass, I bridge the 2 calls. All this works fine until the calling person hangs up before the call is bridged. It sends my CPU to 99% and TOP shows its asterisk doing this.

It might be my Answer() or Hangup() or even my MacroExit() being in the wrong places, but why wont asterisk dump the call immediately upon the calling end hanging up? The CPU stays at 99% until the called end finally hangs up the call. Any help would be GREAT.

[Inc-call]
exten => _X.,1,Answer()
exten => _X.,n,Background(state-name|m)
exten => _X.,n,Read(userdest|enter|10|n|1|7)
exten => _X.,n,Record(/voicefiles/TEST/${userdest}:gsm|3|3|s)
exten => _X.,n,Dial(SIP/+1${userdest}@route-3||Cm()M(test-dial))
exten => _X.,n,Hangup()

exten => h,1,Hangup()

[macro-test-dial]

exten => s,1,Background(/voicefiles/TEST/${userdest}|m)
exten => s,n,Background(is-trying-to-call|m)
exten => s,n,Background(to-talk-to|m)
exten => s,n,Background(/voicefiles/TEST/${userdest}|m)
exten => s,n,Read(accdeny|and-accept|1|n|2|4)
exten => s,n,GotoIf($["${accdeny}" = "1"]?20:100)

exten => s,20,Background(invoice|m)
exten => s,n,MacroExit()

exten => s,100,Set(TIMEOUT(absolute)=1)
exten => s,n,MacroExit()

exten => h,1,MacroExit()

Statistics : Posted by valentine77 • on Thu Feb 09, 2012 1:15 pm • Replies 0 • Views 10

Hi

And the log file shows ??

Drops on ISDN can be many thing , loss of sync etc.

Ian

Statistics : Posted by ianplain • on Thu Feb 09, 2012 1:12 pm • Replies 1 • Views 20

directmedia is just a renaming of canreinvite, as there are other reasons for reinvites, so the original name was misleading.

I note that it says "X-asterisk-Info: SIP re-invite (External RTP bridge)" which means it is due to the use of directmedia.

The channel name doesn't look right. I would expect SIP/GW1G729-........., although maybe this is a 1.8 thing. Previous versions would have used the target address when not using sip.conf, and the sip.conf section name, when using it. Are you doing that? (putting the target address in sip.conf, rather than referencing the sip.conf section name? That would cause you to get the default directmedia setting.

Generally including the dialplan CLI output from, at least, verbose level 3, makes it much easier to see what is going on.

Statistics : Posted by david55 • on Thu Feb 09, 2012 11:50 am • Replies 8 • Views 222

Thanks for the update.

Statistics : Posted by ctdw • on Thu Feb 09, 2012 11:31 am • Replies 2 • Views 32

Monitor will frustrate directmedia. Maybe you have an inappropriate use of directmedia?

Statistics : Posted by david55 • on Thu Feb 09, 2012 11:20 am • Replies 1 • Views 14

Hi - this was indeed an issue found in 5.1.2 (and 5.1.3) and has been fixed in our upcoming release (5.5) code base. Unfortunately, until 5.5 is available, you'll have to use an alternate browser on Mac.

Statistics : Posted by barrangatan • on Thu Feb 09, 2012 11:10 am • Replies 1 • Views 16

OK, so I've got an extension like: exten=>_X.,1,Dial(SIP/${EXTEN}@provider-out) and there is no voice when I make outbound call. But when I add a Monitor function before Dial like this:
exten=>_X.,1,Monitor(wav, blah)
exten=>_X.,2,Dial(SIP/${EXTEN}@provider-out)
everything works just fine . Isn't it strange? And does anyone has any ideas how to fix it so I could have voice even without Monitor function?

Statistics : Posted by MrMario • on Thu Feb 09, 2012 11:08 am • Replies 0 • Views 1

Although CEL is generally the solution to non-trivial logging of calls, CDRs should should hangups on the queue. You may need to enable logging of unanswered calls.

Statistics : Posted by david55 • on Thu Feb 09, 2012 10:47 am • Replies 4 • Views 27